Kevin, if I use ulaw for my sip users and my sip providers.... will i minimize the "transcoding" hit to uncompressed mode to my server?
or will the load be the same even if I use g729 everywhere? Im trying to optimize my setup as to do transcoding/uncompressing to a minimum. On 6/3/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
----- Erick Perez <[EMAIL PROTECTED]> wrote: > Or if i have SIP/g729 users and i create a conference with other > users > also at g729 asterisk will not transcode (when using app_conference)? It is not possible to mix conference audio together without converting it to an uncompressed form first. app_meetme in Asterisk 1.2.x certainly does do more transcoding (both inbound and outbound) than is absolutely needed, which app_conference does not do. However, app_meetme in SVN trunk (soon to be Asterisk 1.4) tries to minimize the amount of transcoding by avoiding the decoding of incoming audio from channels that are not speaking and by re-using the transcoded output for channels that share a format (codec). This should make it perform similarly to app_conference in many situations. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
